shopcomputer
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Posts posted by shopcomputer
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This requires some major changes, including new audio prompts. This will take some time.
I have a customer looking to switch to our hosted system, with his current provider if someone mistakenly goes to his extension and it comes to his cell phone, he hits ## and the extension number to send him to the correct extension. He has no audio prompts.
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cant wait to have that option to transfer back from the cell phone to an other ext,
any idea when this will be available ? 4.3 or 4.4 ? i hope not at 5.0
thanks
Any update on this?
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This error is not coming from Snom ONE, it is coming from IIS, you must have iis listeining on port 80, causing PBX to grab a different port.
After using version 2 and 3 for years without hassle, now I have trouble to use version 4 so I am kept on version 3. Now with Snom One I have similar error messages about the OLE registry when I try to access the localhost with downloaded ver.4
Is there any pre-requirement on a Windows computer? I have tried IE and Firefox.Here is the message I get:
error '8002801c'
Error accessing the OLE registry.
/iisHelp/common/500-100.asp, line 17
Any advise?
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I need a list of SIP phones that support video calls. I also need a list of SIP phones that have a web browser or XML browser capabilities. Input would be greatly appreciated.
Video
Grandstream
Cisco
Polycom
XML
Snom
Polycom
Aastra
Cisco
Grandstream
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Any way to set up the wake up call for other ext in the same domain ?
here is the scenario its an hotel and the main front desk would like to be able to set up the wake up call for all the rooms, such as *62*602 etc
is there any way to do that ?
please advice
They should use the interface or SOAP.
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Does a Certificate need to be updated from time to time?
Yes, when it expires.
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I have a sipjack with a trunk, registered to my hosted PBX as an extension.
When the sipjack forwards calls to cell phone, it shows the ANI set on my hosted PBX, I am trying to get it to display original callers ID, depending on the settings I choose, my hosted PBX either rejects the call as an unknown call without trunk info, or it shows the ANy set on the extension, if no ANI is set it shows the extension number.
Did anyone play with this, and get it to forward ani from the sipjack?
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Hello,
We have a company that faxes price sheets to their clients weekly. Probably about 40 clients and each price sheet is 15 pages.
vitelity seems nice but you can't seem to email/send a fax to a distribution group, you need to sent a seperate email to each fax number.
Maybe I'm missing something?
Anyone worked through this senario? Ideas welcome.
Matt
We use vitelity for faxing however fax broadcasting is not their intention.
We have used GFI fax software the last time we had to set up broadcasting.
There are many fax broadcasting services on the web now, jblast comes to mind, google fax broadcast.
You also may be able to do something with vitelity's api.
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Sipjack does not save the DNS settings, when manually specifying an IP address. 4.0.1.3499
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what does hazard symbol in registration status mean?
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allright... but why would you say that... what would isa not do that any other firewall will do?
ISA opens packets and is not application aware of SIP, at least till last version we use, which is ISA 2004. we had to get rid of ISA at our clients that went to sip.
I do have 1 client where it started miracliously working, I won't touch it.
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You can use * replacement *This is my present dial plan on my asterisk's box1NXXNXXXXXX
1+NXXNXXXXXX
1402+NXXXXX
I apparently do not understand how to add a dial plan in pbxnsip
If I try to delete the 100 in the first field and use any one of these as a pattern it does not take.
What I want to accomplish is the user being transparent to the switch over when we make it.
They pick up the phone, dial the number, and get connected.
If they dial a 7 digit number it is knows the area code.
If they dial 1+10 digit number it is a ld call.
They do call Canada on occasion but International is not allowed.
Thanks in advance.
You can also xxxxxxx replacement 1402*
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Hi all,
We've got 3 employees working from home which I want to give the facility to register to their extentions on pbxnsip. We've got static IPs, so I'd like to know is there a guide on how to achieve this? We use ISA server as our firewall and proxy and I have a static IP available for use just for sip and rtp traffic.
Could you give me some sort of guidance on what I would have to set on pbxnsip to get the registration and rtp traffic flowing correctly? I tried this once when we still on dynamic ips and I could get the phones to register, but I couldn't get any voice once a call was established. Any ideas?
Regards,
Rudi Groenewald
ISA is not going to pass sip traffic.
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Moshie,
I would appreciate the Vitelity config.
Thanks for your response.
Chuck
At vitelity, have them route sip to your PBX IP address. then create 2 trunks 1 for inbound 1 for outbound, see attached
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I am a novice and hoping to learn enough about pbxnsip to possiably become a reseller.
I thought the best way would be by setting up the demo to test my ability to understand the application.
My experience is that I rent a virtually hosted trixbox application from one company and then get my did's from another company called Vitelity.
I contacted Vitelity and asked them if I could use pbxnsip with their service which they said I probablly could, but they do not have a config for this pbx.
Looking thru the fourm the only USA provider I see is nexsip. I have contacted them but have not had a response to my request.
Are there any other sip providers that support pbxnsip, that are reasonable, and have configs for this application?
If I have posted in the wrong area please forgive my lack of knowledge. I look forward to being "schooled" and being a good citizen in this forum.
We use vitelity as one of our providers, I will pladly share a config if you still need it.
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Can anyone recommend a suitable hardware to regenerate signal, what should I buy if I must be running for 250 ft?
Kind of urgent please
regards
250ft is fine, 100 meters = 328ft. to extend after that just put any network switch, (with poe if needed) in the middle. There are also extenders avilable for sale, see http://www.ippbx.us/Home/tabid/408/txtSear...er/Default.aspx
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We're trying to install it on Windows 2008 server R2
What is the parameter for the no-daemon?
We are running successfully for quite a while on 2008 R2 64bit.
Is the service not starting? Or you can't get in to web interface? make sure nothing is conflicting with your web port.
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Put the address of the service provider into "Explicitly list addresses for inbound traffic", for example "12.32.12.32" or a DNS A/AAAA name like "sip.proxy.com".
Just upgraded again and put the address in the "Explicitly list addresses for inbound traffic", still global trunk was not finding inbound users, so I downgraded again to 4.0.1.3452 (Win32)
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Hi,
We have a software called CallScout developed for PBXnSIP which I believe serves your needs.
The link for the demo is: http://callscout.vertical.com.co/callscout
DEMO OVERVIEW
Demostration data corresponds to a fictitious company called Demo. This company is equipped with a Pbxnsip IP PBX, which has two domains defined: Demo Sales and Demo Research. Each domain receives and sends calls through a set of trunks. Demo Sales has two trunks, called Local and Global, while Demo Research has also two trunks, called PSTN and International.
Demo Sales domain has three extension accounts: 2010, 2011 and 2012, and Demo Research comprehends the extension accounts: 2001, 2002 and 2003. ACD agents can login using any of these extensions, in order to catch calls received on Agent Groups. Both domains, Demo Sales and Demo Research, have an Agent Group defined, with account number 100.
USERS
The following demo users belong to each one of the existing roles and are available to access the application:
Enterprise rol:
Has access to all domains, trunks and extensions of the Demo enterprise.
Login: demo-enterprise
Password: demo
Domain rol:
Has access to all trunks and extensions defined for its assigned domain.
For domain Demo Sales: For domain Demo Research:
Login: demo-sales Login: demo-research
Password: demo Password: demo
Trunk rol:
Has acces to calls coming into and from its assigned trunk.
For trunk PSTN of domain Demo Research:
Login: demo-research-pstn
Password: demo
Extension rol:
Has access to calls made from and coming to its assigned extension only.
For extension 2001:
Login: demo-2001
Password: demo
Pricing?
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2010/04/23 07:18:31: Global trunk INBOUND 31@localhost could not identify user for 8012149000
rolled back to 4.0.1.3452 (Linux)
and its just fine.
Global trunk vitelity in@pbx.company.com could not identify user for 7185696635 same here, running win32.
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The other option might be Microsoft Exchange 2007, which also supported at one point (?) fax to email. If they already have it, they might just need the installation service to get it going.
They are using Exchange 2003
, . No luck there. -
But you hear the prompt that asks you to connect?? Does DTMF work otherwise when you call the cell phone? For example, when you have a mailbox message, the system calls the cell and then you can listen to your mailbox messages and delete them with DTMF?
Yes, I hear the promps however it does not respong to the prompts.
I set it to call me when voice mail arrives, it does not either respond to 1 to pick up the message.
However, it I call in from my cell phone to the system, it fully responds, lets me place an outbound call, and puts through dtmf's on that call to a remote auto attendant. worked with 4.0.0 stopped working with 4.0.1.
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If the fax server uses the call-ID in the To-header, then it should be okay to use just one extension with 32 alias names.
I think there was a French company offering fax to email. There was already a discussion in the fax forum some time ago.
Fax is not a simple topic, it is not that we just do a quick hack over the weekend and have a stable fax solution.
Need some guidance, would love to get this 80 extension sale.
His other option is upgrading his current Telrad system for 20K to support ip extensions, the upgrade has unified messaging including fax. I know PBXnSIP does not support it directly.
I am looking for a cheaper or even open source option, rather than Faxback, and a methed to route all fax traffic or Fax DID's to that other system, with using 32 licenses, I guess creating some sort of trunk.
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I need some help with Fax to email for a prospect, I am not to familiar with this scenario as I usually use outside fax services.
Client will have 80 voice DID's and 32 fax DID's coming in on a T1/PRI, they want IP PBX for the voice and faxes to come to email.
I was going to go with, a sangoma card to bring the T1 into a PBXnSIP server, then how do I get PBXnSIP to forward the 32 fax DID's to a fax system?
Do I need to purchase 32 extra extensions on PBXnSIP and create 32 autoattendands to forward those extensions?
Anything less expensive than Faxback, I called them, it cost something like $3200 for 4 ports?
http://www.pbxnsipsupport.com/index.php?_m...kbarticleid=661
, . No luck there.
registration with Snom One
in Polycom Phones
Posted
I migrated a customer from PBXnSIP to Snom one Blue, the majority of their phones are Snom and re-provisioned succesfully.
They have 2 Polycom 335 phones that I must get working on the Snom one, how can I get them provisioned. I even tried manually in the web interface changing the IP, user name and password and it does not register.